Questions tagged [dsp-core]
**Do NOT use this tag for general questions about digital signal processing!** A DSP core is a processing unit specialized for digital signal processing tasks.
23 questions
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Intuitive explanation of cross-correlation in frequency domain
According to the cross-correlation theorem : the cross-correlation between two signals is equal to the product of fourier transform of one signal multiplied by complex conjugate of fourier transform ...
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3
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Get a N-FFT with two N/2-FFT already computed
After somme researchs on the web, I don't find the answer of my problem (or I don't understand it) and I hope this post will succeed.
I'm working on a real-time FFT convolution algorithm (C++) which ...
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Mathematically Inclined Signal and Systems / Signal Processing Book Recommendations
I'm an electronics engineering student with high inclination to analysis and pure mathematics. I was just wondering if there was any book ( or any resource ) that treats signal and systems and signal ...
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How to pick coefficients for Fractional Delay Filters?
I have a Virtex 6 FPGA running at 200MHz with ADC/DACs on it. I have been converting a WiFi signal (2462MHz) down to a more reasonable IF of 25MHz, sampling, running the signal through taps, tweaking ...
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Raised-Cosine Filter - calculating phase response
I want to compute phase response of an raised-cosine FIR filter. I have trouble finding the complex transfer function of the filter in literature (there is a real valued transfer function of frequency)...
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Costa's loop questions, design, tracking and limitations
Why does Costa's loop PLL bandwidth need to be narrow? and why can't it track a certain frequency offset? and does that hold even if it has a filter with a pole at DC?
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2
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Hardware kits for DSP Lab?
I have been assigned responsibility as Lab instructor of DSP at undergrad Level and so i have to equip my lab,please recommend which processors should i purchase for hardware implementation of dsp ...
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2
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Best method to do piano key pitch detection?
For example, when playing piano, at the same time, print out the key notes by analyzing the signal of piano sound.
How to do some pre-processing to remove the noise? When calculate with FFT, noise ...
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How to decide what platform is best to implement real-time audio processing on?
When developing products I do all of my algorithm design in Matlab. They are usually pretty basic, an IIR or FIR filter or two, a couple FFTs, etc. When it comes time to move it to an embedded ...
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What does 'canonical' mean?
What does 'canonical' mean?
e.g. 'write IIR or FIR in canonical form'
I get the gist of it but maybe someone here has a precise definition. In a non-DSP context it would mean something like '...
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SNR of a 16-bit DSP with 12-bit ADC, 40-bit accumulator
According to the 6dB-per-bit rule, a 16-bit DSP would provide 96 dB of dynamic range, approximately.
If the ADC and DAC are both 12-bit long, the DSP is still considered to have 96 dB, or the ...
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OFDM training symbol format
For OFDM synchronisation, we are using two training symbols according to the Schmidl and Cox method of frequency synchronization and timing synchronization. The paper says:
The first OFDM training ...
- 1,052
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1
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What's the difference between using DFT, IDFT or DCT to calculate cepstrum of a power spectrum?
I've seen different equations that calculate cepstrum from power spectrum, but the equations are not consistent. Some people use Fourier transform, some use the inverse Fourier transform, and some use ...
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Hardware requirements for ambient active noise cancallation system using adaptive LMS
Background
I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ...
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FFT of a $N$-length real sequence via FFT of a $N/2$-length complex sequence
I am doing FFT. The one way I did is to put real part in one array and the imaginary part as zeros and then calculated the FFT. But this take up lots of resources.
Now I want to calculate FFT by ...
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Estimating the amplitude of a particular frequency
I am relatively new to DSP and have been reading a lot on the internet. I have a couple of questions.
I have a signal in the form of a function
$$
f(x) = A_0 + A_1 \cos(\omega_1 x) + A_2 \cos(\...
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Hilbert Transform, filters - two different phase graphs
Relationship between magnitude response and phase response for minimum phase is:
Phase plots are obtained from Hilbert transform of real magnitude values. Input to Hilbert transform was a vector (...
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1
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How to prove energy is preserved in sampling using Parseval's Relation?
I have a continuous signal,xc(t) which is bandlimited to =pi/T ,
and x[n] = xc(nT) are samples of this signal.
I need to show there is a Parseval's relation between the original signal and the sampled ...
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Computing phase difference between two sinusoidal signals
Based on this post here, I tried to calculate the phase difference between sine waves. Here is my code.
...
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FFT - Effects of Windowing,Improvisation approaches on RealTime FFT Hardware
Fore-word :Kindly Observe this question has multiple updates take the latest update as question !
here below is my 256pt FFT on an 2Khz sinusoid signal,on a fixed point DSP tms320c5515
please ignore ...
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Speech Activity Detection- Adhoc Problem with multi source sounds(birds,trees,heavy air) and how to eliminate them completely
Here i will first try to put my application agenda
Typically i am sampling at 8Khz rate and filtering speech components using a BPF(digital)
now i have got the 0-4Khz components in which ...
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0
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the number of multiplications per second required by the system
Q1. Consider a system implementing a rational sampling rate change by 5/7: for this, we cascade upsampler by 5, a lowpass filter with cutoff frequency π/7 and a downsampler by 7. The lowpass filter is ...
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FIXED POINT conversion Problem (DSP)
Actually after reading the data from ADC is an $Q15$ format (ie. ADC supports 0-3.3v, ADC is an unipolar type). then the $Q15$ data is converted to $q31$ format ($Q31$ value = $q15_{data}$ <<16)....
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